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    Voice Over IP.doc Voice Over IP.doc Document Transcript

    • Nexus VoIP White Paper VoIP Presented by Nexus Management Nexus VoIP White Paper 1 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Nexus VoIP White Paper INDEX: Introduction: In most companies two isolated communication infrastructures exist. Local Area Networks (LANs) and Wide Area Networks (WANs) are used for data communication and Telephone (PBX) networks for voice communication. The introduction of two new communications technologies based on the Internet Protocol (IP*) make it possible to converge traditional telephone and data networks into a single system. For users who have free or fixed-price Internet access, Internet telephony software can also provide free inter-company telephone calls anywhere in the world. To date, however, Internet telephony does not offer the same quality of telephone service as direct telephone connections. There are many Internet telephony applications available as well as a number of IP enabled telephony appliances. Some come bundled with popular Web browsers. Others are stand- alone products. This not only provides new application options for business communication, it also offers companies a significant cost-savings potential. This white paper is a description of a diagram and details of a complete VoIP using an Asterisk** PBX and communications gateway. Extensive modifications have been made to qualify the end product as a secure and robust, commercially viable Telephony alternative: Nexus VoIP White Paper 2 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Nexus VoIP White Paper a) Integration of a comprehensive VoIP platform: a. Provisioning b. Billing (in development) c. Secure Instant Messenger (Jabber***) d. Analysis (CDR) e. Public Demo / Internet Gateway (in development) b) Help customers achieve VoIP using the Nexus Data Centre services. This works in two modes; a. A complete PBX b. A VoIP gateway and bring them onto the Nexus network. They can then benefit from the items below: c) Use a 3rd party VoIP carries to carry calls all over the US at less than Telco pricing. Any customer could then make calls to anywhere in the US at very competitive pricing: With a VoIP, Nexus can offer better than Telco rates. In the future, SIP**** to GMS devices may well become available that allow a call to go from VoIP to GSM and save the cost incurred when a call from a Land line to a GSM provider is made, as mobile to mobile calls tend to be cheaper than land line calls. Nexus VoIP White Paper 3 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • The Classical Telecommunications System Classical telecommunications systems are in most cases proprietary systems of the respective provider. Therefore, it is only possible for manufacturers to develop and sell expansions for hardware and applications. It is also impossible to operate a system phone of one manufacturer together with the system of another manufacturer. In addition, the simplest applications, such as Music on hold, which is played while a caller waits, or an Auto Attendant, which answers calls with automatic welcome texts, require additional expensive modules, installed by the manufacturer. Even if one just wants to change the melody or the announcement text, the manufacturer’s support usually is required. The IP-PBX Concept The IP-PBX is a completely new concept for a telephone private branch exchange. In this concept, the voice data is not directed through a separate infrastructure but it rather becomes an integral component of a common infrastructure for both voice and data communication. This integration is based on the packet-oriented communications protocol IP (Internet Protocol), which has begun its triumphant advance through the communications world as a result of the rapid development of the Internet. Here the voice data is compressed and simultaneously transported in the form of IP data packets together with other types of data and using the same network. The IP-PBX itself is a software package running on the Linux platform. This software package controls all dialing and connection procedures within the local (and non-local) IP network. The essentially new feature of an IP-PBX is the fact that no special “switching hardware” is required. Therefore, there is no switching unit as is the case in a classical telecommunications system. The IP-PBX provides the call handling functions of a telecommunications system by assigning IP addresses and phone numbers to the IP telephony terminals connected to the PC network, and by appropriately coordinating and controlling the connection requests. Due to the fact that not only connection control data but also voice, fax, or video data are transported as IP packets, the IP-PBX retains complete control over existing connections at all times. Therefore, it is possible to dynamically integrate voice and data applications into a uniform, total system. During connection control and depending on current data structures the IP-PBX can easily create, play, or record voice information, for example. Furthermore, it can function as an automatic announcement Nexus VoIP White Paper 4 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Reservations about using IP Telephony Telephony is surely one of the most important mission critical applications in a company and therefore it must satisfy the highest quality requirements and fail- safe criteria. It is therefore absolutely necessary for IP telephony to measure up to the high standards of classical telecommunications systems. Voice Quality In general, the quality of voice transmission in a network is influenced by various factors. Technically, voice quality in the IP network is particularly dependent on the following factors, some of which also demonstrate dependencies to one another. 1. Voice coding and compression (Codec) 2. Packet delays (latency or jitter) and 3. Packet loss during voice transmission (particularly when delivery is “too late”) The most important progress made in packet-oriented voice transmission within the last years has been achieved in the area of Codecs, which scan the voice signal with the help of special algorithms and then compress and pack the signal into data packets. Another factor, which affects the quality of voice transmission, is packet delay. The “delay” factor does not only occur within the Codec; it is also triggered by different influencing factors on the transmission route: • Buffering • Queuing • Switching/ Routing • Fixed transmission time • Long-distance trunk transmission time Audio Coding In order to handle real-time-oriented voice applications via packet-oriented Internet Protocol, it is necessary, to compress the data to be transmitted. For this reason, the ITU (International Telecommunication Union) has created a number of standards, which represent a variety of different voice qualities, depending on the available bandwidth. Delays are caused in one way or another by the time it takes to process the packets and by their temporary storage. Greater throughput performance and an efficient storage concept will minimize the influence of these factors. The more routers in the IP network a voice packet must go through and the longer the Nexus VoIP White Paper 5 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • temporary storage times in the individual connection segments are (for example, due to heavy data traffic), the more the delays will accumulate. At the same time, packet loss can never be fully eliminated in a data network (due to transmission interference or router congestion). When transmitting data, this problem is alleviated by the fact that the packets sent can be temporarily stored and, if they are lost, it is possible to repeat the transmission. This procedure, however, is not suitable for IP voice transmission. A packet must be received within a specific time frame in order for the algorithm to function correctly, which converts voice data packets back into analog voice signals. If a packet is received “too late”, it cannot be used for the Codec and is considered lost. This problem can only be solved for good in circuit switched networks in which there is one completely reserved transmission route for every connection – just like in a classical telephone network. Fortunately, the perceived voice quality is not significantly affected when there are modest packet losses; a loss of up to 5% is almost imperceptible. It can be assumed that even private telephone users will only accept poor voice quality in exceptional cases, in contrast to the GSM mobile telephony technology, where the often poor voice quality makes up for the advantage of being reachable at anytime. Due to the fact that voice communication is a particularly critical element of the business world, service providers and VoIP system providers will have to demonstrate and guarantee “Quality of Service”. A number of protocol elements and procedures assign voice packets a higher transmission priority than data packets. This not only reduces delays and latency, but it also prevents the complete loss of packets. IP in Public Networks A crucial problem in public networks such as the Internet is the provision of sufficient bandwidth. Since 1998 IP network have developed “an abundance” of bandwidth by extending the capacities on classical long distance connections by a factor of 1,000 to 10,000. This will be achieved by implementing new fiberglass routes. As a result, the bandwidth required for IP voice connections will more or less be guaranteed because they only require a very small bandwidth of 8 Kbytes/s each. This effect can only be fully exploited for telephony in the “public Internet” when the majority of ISPs (Internet Service Providers) and carriers implement this new technology. Once they do, Internet telephony will also achieve an acceptable voice quality range. Nexus VoIP White Paper 6 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • IP in VPNs The situation is much better in companies which already have their own network (Intranet) or a special Virtual Private Network (VPN). In these networks, it is possible to achieve very good voice quality using currently available technology. The voice quality in these VoIP networks is not any different from that which is provided by a classical telephone network. IP in LANs The quality considerations for local area networks (LANs), which will play the most important role in the transmission technology for IP-PBXs, appear to be even more positive. The VoIP technology will benefit from the fact that the old 10 Mbit Ethernets are currently being replaced by 100 Mbit Ethernets in LANs and even these will soon be replaced by 1 Gbit Ethernets. Furthermore, networks in companies today are already highly structured with hubs and switches, so that there are seldom more than 50 terminals connected in a segment which must share a bandwidth when communicating simultaneously. If one assumes that a maximum load of 10% of the available bandwidth is required in order to guarantee the “delay-free” transportation of voice packets even during simultaneous data traffic, then a 10 Mbit Ethernet can now transport between 25 -100 voice connections parallel to any other data traffic. The Structure of an IP-PBX Network Four components are generally required when setting up an IP-PBX infrastructure: 1. The telephony client. This can be made available in the form of an IP telephone or as software on the respective PC. 2. The IP-PBX router/server is the central element of the infrastructure. It is responsible for all connection requirements and call handling processes. 3. An Integrated gateway, necessary for providing the connection between the various network structures of the IP network and the public telephone network. 4. In addition to this, multi-location installations must be implemented. Nexus VoIP White Paper 7 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Some typical VoIP network solutions: Cheap and Simple Intra-office: shown here as a three office link, but there is no restriction to the number of links Stockholm Normal analogue Phones Nexus VOIP box Tokyo Sydney 3 way point to point Nexus VoIP White Paper 8 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Components of an IP-PBX 1. Server The heart of an IP-PBX architecture is the IP-PBX server, which is responsible for all call handling processes and connection requests. It is also responsible for the specific redirection of calls if the line called is “busy” or “not available”, for conference calls. Because the telephony server processes all connection requests, it is also able to keep track of statistical information concerning the utilization of communication paths and the condition of all current connections, and it contains the status of all registered users. The actual voice data, however, is directly exchanged between the telephones and it will only be directed via the server in exceptional cases – such as for a telephone conference, for example. Furthermore, the server is responsible for special features. The capability of an IP-PBX is particularly evident here. These features include functions such as: • Hold • Redial • Call Redirection • Speed Dialling • Call Waiting • Conference calling • Information on charges which, together with their integration into existing or new PC applications (Outlook, Exchange, Lotus Notes). In addition to these basic functions, a multitude of special functions can easily be implemented, including the following classical telecommunications system functions: • Direct announcement (“Will the driver of the car XY please clear the driveway” or “Ms/Mr Z, could you please bring me the XY file”) • Call redirection if “busy” or “delayed”, either internally to another employee, to a group, to an answering machine, or externally to a mobile telephone (“Do Not Disturb”). • Call swapping between two calls (both internal and external calls) and connecting these calls to one another or advanced features, such as • Group Function: If the person called does not pick up the call, the call will either be displayed within a group (for example Sales or Accounting) or it will be picked up by the “Auto Attendant”, which offers the caller several options for handling his call. • Personal Answering Machine, voice mail 2. Gateway Nexus VoIP White Paper 9 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • The gateway in an IP-PBX architecture acts as a bridge for the communication between IP telephones and the classical telephone network (PSTN). A gateway used exclusively for connecting local IP telephones and PC clients usually consists of one or more commercially available cards used to terminate T1, E1 and BRI connections. Calls, which are made from the local sector to distant IP addresses, use one of the compression standards supported by the respective telephones. The gateway can also create a connection to an existing telecommunications system by using an appropriate protocol extension. As a result, it is possible to use the IP-PBX not only as a substitution, but also as an extension to the existing telecommunications installations. 3. Software Client By using client software, a normal PC workstation, which is equipped with a sound card, speakers, and a microphone (a typical multimedia setup these days), can be transformed into a high performance LAN telephone. Depending on how the user would like to equip his PC, it is also possible to use a headset or a handset. The client is equipped with a graphic user interface (GUI), which is operated with the PC mouse and which is usually displayed in form of a telephone with special features. Advanced telephony clients are provided with the option of configuring the user interface in terms of the design and range of functions any way you want. This configuration can correspond to company or department standards, or be adapted to meet the needs of the individual users. This way, for example, any user can vary the number of, the assignment, or the position of speed dials in just a matter of seconds. There are no limits to the design options here. Such design flexibility is impossible with classical telephones: There is a FREE soft phone from X-Lite, a premium SIP softphone with many PBX-like features. Open standards-based design allows for maximum network interoperation and integration. Nexus VoIP White Paper 10 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • X-Lite Features: • Touch-tones [DTMF] • 3 Lines • Multiple Proxies • Line Hold • Inbound Call 'Ignore' • Inbound Call 'Go to Voicemail' • Call Forwarding URI/URL • Voicemail URL • Dial/ Redial/Hang-up • Dynamic CODEC Selection • Caller ID [SIP ID] • Call Timer • Silence Threshold • Backspace/Clear/Delete • Mute • Microphone & Speakers Levels • Microphone & Speakers Meters • Push-to-Talk [PocketPC] • Last Caller-ID • Recent Calls Dialled • Recent Calls Received Nexus VoIP White Paper 11 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • • Sound Device Selection • Direct IP to IP Calling • New Familiar-looking Menu • Phonebook [Import/Export CSV] • Speed Dial Benefits: • Easy to install • Intuitive user interface & Menu. Easy to navigate and configure. • G.711/SPX/iLBC/GSM Codecs included • Speakerphone • Standard PC and PocketPC hardware • NAT/Firewall support • Specify NAT IP to be written in SIP messages • Supports Windows 98SE/NT4/ME/2000/XP • Supports Mac OS X 4. IP Telephone An IP-PBX should not only be run with a software client exclusively, because there are some cases in which an additional desk phone may be required or a PC telephone is not available. In these cases, it is also possible to operate an IP-PBX with an H.323-compatible telephone. The only difference between an IP telephone and a classical telephone is that the IP telephone uses a different interface: Instead of an ISDN interface, the IP telephone has a 10/100BaseT interface. Of course, the protocols that run within the telephone are also different. In addition, IP telephones also provide part of the necessary voice compression. The type of IP-PBX to be implemented depends on the corporate structure. The simplest configuration is the single location solution. In this case, the company has one central server and one central gateway. While the central server supplies all of the necessary user information, number schemes, etc., the gateway is responsible for creating a connection between the company and the classical telephone network or other IP networks. Even if the IP-PBX must be connected to an existing telecommunications system, this will not result in any problems for the IP-PBX hardware. An IP-PBX is often implemented as an extension to an existing telecommunications system. At first, only several individual departments are equipped with an IP-PBX as a sub-system of the existing telecommunications system. In addition, external locations, such as home offices, can be linked to the company network using existing IP connections. Nexus VoIP White Paper 12 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • If IP telephony proves useful in a company, successive new connections will be configured directly over the IP-PBX. The old telecommunications system is gradually replaced by an IP-PBX. This step-by-step introduction of an IP-PBX allows companies to identify and, if necessary, to solve problems which may arise concerning the utilization of the network or fail-safety issues in the early stages of the migration. Stand alone VoIP products at the moment are more expensive than normal phones since they are technically more complicated and produced in much lower numbers. Since VoIP phones are IP devices, they are not only physically compatible to the computer network. They actually are small computers and thus can interface on the application level - this is called CTI (computer telephony integration). IP Phones have an integrated web interface to make configuration and remote management easy. The graphical interface is there to make using supplementary services (hold, transfer, divert, conference, etc) easier. Example: A phone from the German company SNOM: An IP telephone with the latest VOIP technology, a two-line LCD display, alphanumerical caller ID and user interface with a multitude of features. This phone supports the Codec G.723.1. And this means compatibility with considerably more components of other manufacturers. With respect to speech quality, even with low bandwidths, this phone can achieve a standard that was previously only possible with broadband technology. Nexus VoIP White Paper 13 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Main Features • Graphic display (two-line) • SIP • 3 dynamic soft keys • Support for different languages • 5 programmable function keys (National Language Support) • Dual Ethernet connection • Supports G.723.1 and G.729 a/ b • Headset connection • Security (SIPS, SRTP) • STUN, UPnP, ICE • NAT support Using IP-PBXs Sensibly The previous pages have primarily dealt with the technology and the structure of an IP-PBX infrastructure. The following pages present the specific applications of such an infrastructure. Unified Messaging The IP-PBX represents a fundamental component of a “Unified Messaging” system. The messages from different applications, such as voice mail, fax and email, which used to be stored separately, are now combined at a common location. In previous implementations in the area of communications servers, the area of voice communication has been extremely neglected or it could only be realized with a great deal of technical effort. The use of an IP-PBX makes it possible to elevate this concept to a new level. It is now possible to handle both voice and data communication on a single client. Employees can log onto the company network with their mobile phones if they are on the road. This enables them not only to access all stored messages, but also to create the voice connections which result from these messages. This is not limited to listening to traditional voice mails or to sending new ones: Emails or faxes can also be “read” to the user via the IP-PBX, if such a function has been implemented. The implementation of this function is quite simple because only a new software module must be installed on the server. At the present, most incoming voice mails can only be checked with the help of a telephone and, due to the lack of a connection between the telephone and the PC, they cannot be processed any further. However, the employee who uses voice and data integration is able to manage all of his incoming messages from his PC under a single interface. Voice mails and faxes are listed in the same Messages-Received list as emails and they can also be treated in the same way. This integration of different communication paths offers companies a productivity improvement that should not be Nexus VoIP White Paper 14 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • underestimated. Individual Telephones Similar to PC applications, a modern system telephone provides a multitude of functions, which an employee will only use occasionally or maybe not at all. In addition, different departments in a company will have different requirements for their telephones. A classical system telephone is not able to meet this variety of requirements because it offers the same number of functions to all users. In addition, if all of the speed dials have already been assigned on a classical system telephone, it is only possible to extend the number of buttons by purchasing a new component. IP-PBXs, on the other hand, offer a degree of flexibility that has never been seen before. This is possible because you do not need a system telephone for an IP- PBX. On the contrary, software, which displays all the functions of a system telephone on a PC, can be installed on an employee’s PC. Advanced clients can design the interface any way they want. This allows an employee to activate only the functions he really needs. If it is necessary to extend the range of functions, employees can add new abbreviated dialling buttons, for example, in just a matter of seconds. All the employee has to do is to use “Copy & Paste” to copy an existing speed dials and to insert it at the desired position on the interface. Then all that remains to be done is to assign the desired telephone number to the button. Furthermore, functions can be added or deleted individually and it is even possible to change the design of the interface to meet the company’s needs. In this way, company-specific telephones can be created, which reflect the corporate identity of a company, and which take the different communication requirements of the individual departments into consideration. The IP Telephone as a Message Centre A classical system telephone is not connected to PC applications and, therefore, its voice functions are limited. The IP telephone, on the other hand, functions as an intelligent message centre. Due to the fact that it is connected to the server of the data network, it is informed about all those events and messages, which are normally displayed on the PC. Therefore, when the PC is not turned on, an IP telephone can inform an employee that he has received an email and then show it on the display. This is possible due to the fact that the IP-PBX has access to the email system. If the IP-PBX is linked to the electronic appointment calendar, the IP telephone can also inform an employee that he has an upcoming appointment. In addition, the IP telephone can also be used to access specific web contents, such as stock market or news tickers. Thus, an employee is not forced to keep his PC turned on all the time and, even if his PC breaks down, he will still have access to all of the important functionalities. Nexus VoIP White Paper 15 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Customer Management Another important reason for using an IP-PBX is the positive influence it has on the company’s communication behaviour as it relates to its customers. Customers usually call the central telephone number of a company in order to have their call redirected to the appropriate employee. However, what often happens is that the connection attempt leads to nowhere because the respective contact is not available and the switchboard operator does not know who is responsible for dealing with the customer’s concern instead. According to a study conducted by the magazine Computerpartner, these fruitless calls cost companies DM 3,000 per employee and year. Furthermore, 30% of all calls require a repeat call due to missing information. The intelligent use of an IP-PBX can help create better conditions for more efficient customer management. A modern switchboard is able to recognize when a particular employee is at his desk and who should be contacted instead if this employee is absent. This knowledge is based on two types of information: 1. The information that indicates whether an employee has logged on to the LAN. 2. The information which indicates whether an employee has an appointment (this is done by analyzing his calendar) The switchboard has direct access to this information via the IP-PBX, which means that no detour through a PC is necessary. This use of a common infrastructure allows the IP-PBX to communicate directly with the LAN servers and to get all the necessary information directly from there. Given the event-dependent call redirection function, the situation in which an employee has forgotten to activate call redirection will no longer occur. If the employee is not logged on, the IP-PBX will know who is responsible instead. If the employee is in a meeting, on his way to a customer, or on vacation, the IP-PBX will redirect all incoming calls to the redirection destination, which may be another employee or the mobile phone of the person originally called. Web Call Centre Because IP telephony is based on Internet technology, it can also be used to equip a company’s Internet presentation with a high degree of interactivity and, therefore, to fundamentally improve customer satisfaction. The Internet presence of a company can never replace a conversation with a customer. Many of the questions a customer has regarding a product or a service are too unique and a company simply could not publish all of the questions and their answers in the Internet. Nexus VoIP White Paper 16 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Moreover, Web pages are designed to be international, which means that customer questions could come “from the other end of the world”. Due to the time difference, making contact could be extremely problematic in such a case. However, these restrictions can be eliminated if IP telephony is implemented as a permanent component of the company’s Internet strategy. The company’s Web page is equipped with a so-called “Call me” button. If a customer activates this button, a message will be generated via the company’s Web server and sent to the employee closest to the customer. If the customer has a telephony client, the employee can contact him or her directly. If this is not the case, the customer can enter his telephone number, which will then be dialled by the employee within seconds. The connection is made using the same application used to inform the employee about the contact request. Because the IP connection between the customer and the Web still exists, the employee can answer questions on the telephone and also directly help the customer to navigate through the Web page. Wireless Communication Using IP Another important step in the creation of an integrated business communication system is the incorporation of mobile telephones. This trend is based on the Wireless Application Protocol (WAP). WAP allows users to access applications in a company’s network and to have them displayed on a mobile telephone. Therefore, WAP mobile phones, when used together with an IP-PBX, are able to seamlessly integrate Sales employees, who are often out of the office, into the company’s communication infrastructure. Call redirections can then be set up or changed even when an employee is travelling. Furthermore, employees are able to access their calendar or to check their new voice mails. If the IP-PBX has an interface to the email system, employees can also send or receive emails when they are travelling. Advantages of IP Telephony The advantages of IP telephony are demonstrated at different levels. The most significant advantages include: • Higher Productivity: The integration of applications which were previously separate, such as email, telephone, and fax, under a single roof enables the individual employee to access information more quickly and more directly and this, in return, helps to improve his productivity. • Reduction of Infrastructure Costs: Because voice and data use a single, common infrastructure, all costs having to do with the setup, operation, and maintenance of the voice network are eliminated. This means that companies no longer need a separate telephone network, neither a complicated switching technology nor its own cable distribution systems. Nexus VoIP White Paper 17 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • In addition, companies no longer need maintenance contracts with telecommunications system manufacturers because the IP network and • the corresponding components can be managed by the in-house IT department or “from one source” by the external data network • administrator. Furthermore, practically all hardware purchases relating to an isolated voice infrastructure become unnecessary because the existing hardware in the data sector can now be used for both voice and data. • Features: The convergence of voice and data allows you to simply and inexpensively implement features, which are equal to, or even better than, any of today’s large telecommunications systems. These features range from the simplest implementation of additional functions, such as voice mail, Auto Attendant, etc., to the integration of the Internet into the communication process, such as the installation of an E-commerce presentation including voice functionality (Web Call Centre). • Greater Flexibility in Manufacturer Selection: Due to the use of international standards, the user must no longer rely on one telecommunications systems manufacturer. On the contrary, he can select the components and functions he needs from a broad selection of providers. • Reduction of Moving Costs: Although this point may seem somewhat irrelevant at first sight, the costs associated with moving an employee within the company should not be underestimated. When the previous level of communication capability is not provided on time in the new office, this not only results in lost work time, but it also causes additional costs for the redirection of the voice and data traffic to the new office and for alterations in the user administration. If you have two separate networks, these changes must be made separately. This means up to several days of delay and usually a great deal of work for making changes. This double effort is eliminated if one uses IP telephony because it is usually no more complicated than unplugging the IP equipment in the old office and plugging it back in the new office. This is possible because all user information, for both data processing and telephony, is contained in a single data structure. Therefore, the administrator has direct access to all data relating to the user, be it his login scripts for the LAN or his telephony settings, such as redirection destinations or abbreviated dialling lists. All changes can be made from a single workstation and in one operational step. • Reduction of Telephone Charges: The majority of a company’s calls are internal calls. According to a recent study, the ratio of internal to external calls is 4:1. If the data traffic of the locations is already being conducted on leased lines, these routes can also be used for voice traffic. Further savings can be achieved by using the LCR functionality. On the one hand, this function selects the least expensive provider for external calls and, on Nexus VoIP White Paper 18 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • the other hand, it directs the calls on the company’s own network for as long as possible. This means that a call from New York to London might first be directed to the branch office in London on the internal network and then from there switches to the public network. Summary This guide has shown the options available for sensibly integrating voice and data communication and for making a company’s business communication more productive and efficient. The use of existing international standards creates a climate of innovation and it encourages competition among manufacturers. As a result, a company can invest in this new technology early on without having to worry about coming to a technological dead-end, which is often the case when an investment is made too early. In addition, the necessary technical know-how usually already exists in the company, so that the company can concentrate entirely on using this technology to achieve critical competitive advantages. Therefore, IP telephony is a “must” when it comes to completely exploiting the existing potential of the Internet. On the next page the Nexus VoIP solution via encrypted VPN tunnels is demonstrated, using the Data Centre in Brunswick, Maine, USA: Nexus VoIP White Paper 19 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Nexus VoIP White Paper 20 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • The diagrams may look a little complex, but most of the pieces are in place. Many of the DMZ pieces have been configured. The rest of the diagram shows the data flow. Many of the services can be consolidated onto single/existing units. The call accounting component has yet to be implemented. In VoIP people use Radius servers to record call details; all of the open source Invoicing and Client Management Products use Radius data to formulate their invoicing. Nexus is using Sip Express Router *** (SER), a high-performance, configurable, free SIP server. It can act as SIP registrar, proxy or redirect server. SER and Asterisk both have 3rd party modules that generate Radius data; these then can be sent to a call accounting platform that is presently being tested. The existing Asterisk PBX’s will also need to be reconfigured to allow a web based provisioning engine. This will involve storing the extension and phone data in a MySQL engine, adding a number of web based modules. There is a Jabber to SER gateway, also under investigation. Nexus VoIP White Paper 21 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • Index of definitions: * Voice over IP (VoIP). Internet telephony products are sometimes called IP telephony, or Voice over IP (VoIP) products, a category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls. ** Asterisk is a complete PBX Linux based software. It runs on and provides all of the features one would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, and Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium™. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card. Also supported are the Internet Line Jack and Internet Phone Jack products from Quicknet. Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next generation voice-data integrated networks and existing infrastructure. Asterisk not only supports traditional phone equipment, it enhances them with additional capabilities. Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using Packet Voice, it is possible to send data such as URL information and images in- line with voice traffic, allowing advanced integration of information. *** Jabber is best known as "the Linux of instant messaging" -- an open, secure, ad-free alternative to consumer IM services like AIM, ICQ, MSN, and Yahoo. Under the hood, Jabber is a set of streaming XML protocols and technologies that enable any two entities on the Internet to exchange messages, presence, and other structured information in close to real time. Jabber technologies offer several key advantages: Nexus VoIP White Paper 22 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.
    • • Rapid and indistinct speech • To talk in a noisy, excited, or declamatory manner • A streaming XML technology mainly used for instant messaging **** Session Initiation Protocol (SIP), an application-layer control (signalling) protocol for creating, modifying, and terminating sessions with one or more participants. ***** Sip Express Router (SER). SER is a very efficient and light weight SIP Proxy. It is not a PBX and PSTN/Media gateway like Asterisk, but is ideal for the demo/public system and as an outbound gateway for Nexus. Most of the professional setups use a combination of Asterisk and SER to build their VoIP solutions. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols. Nexus VoIP White Paper 23 NOTE: The information contained in this document is the property of Nexus Management and is disclosed to you on the condition that you maintain the information strictly confidential. You are hereby warned that the information disclosed is subject to change without notice or the assumption of any liability on the part of Nexus Management.